Hello,
Does Asterisk PJSIP support an ice restart from UA ? When i make some
tests, Asterisk reply SDP without ICE candidate and the
setRemoteDescription on my UA failed because there is no ice candidate.
I read the RFC https://tools.ietf.org/html/rfc8445#section-9 and it not
clear for me, i
If canreinvite=yes is specified in sip.conf for 2 sip extensions and call
recording is disabled in asterisk, both legs have same codec . Doesit
always does native bridging . I am
using freepbx . How can i know if a call is going through asterisk or
they are bridged directly to each other ? Does
Hi,
I have a sipphone behind a router doing NAT, an asterisk
box in the middle and another asterisk box, which works
as gateway to further destinations.
The asterisk box in the middle should do all call setup
and tear down, but no RTP. RTP should flow directly between
the sipphone via the router
On Fri, 2006-06-30 at 10:39 +0200, Roger Schreiter wrote:
> Hoa Thai Duy schrieb:
> > Roger
> >
> > If transcoding happened (from G.729 to GSM, any to any) then Asterisk will
> > issue no re-INVITE, for sure.
> >
> > Pls. change
> >
> > Disallow=all
> > Allow=gsm (only one codec)
>
>
> Hi,
>
Hoa Thai Duy schrieb:
Roger
If transcoding happened (from G.729 to GSM, any to any) then Asterisk will
issue no re-INVITE, for sure.
Pls. change
Disallow=all
Allow=gsm (only one codec)
Hi,
yes, to avoid transcoding problems I only have one
codec, just alaw. Anything else is disallowed.
T
D] On Behalf Of Roger
Schreiter
Sent: Friday, June 30, 2006 8:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP reinvite still does not occour
Hi,
I have in my sip.conf
disallow=all
allow=alaw
in order to avoid any codec problems disturbing reinvite.
And of course I
Hi,
I have in my sip.conf
disallow=all
allow=alaw
in order to avoid any codec problems disturbing reinvite.
And of course I have:
canreinvite=yes
In extensions.conf there is only one Dial command. It
has no qualifiers like t or T.
Just Dial(SIP/[EMAIL PROTECTED])
Anyway, asterisk does not tr
ED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Mark PhillipsSent: Thursday, September 15, 2005 7:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP reinvite asterisk and NATIf these phones are all to be in a single location I'd deploy a remoteAsterisk bo
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Thursday, September 15, 2005 7:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP reinvite asterisk and NAT
If these phones are all to be in a single location I'd deploy a r
> If these phones are all to be in a single location I'd deploy a remote
> Asterisk box and run an IAX trunk between remote and local sites.
> That'll save more bandwidth than having a potential 5 individual SIP
> sessions running over your link.
One more potential point of failure, rather not.
>
If these phones are all to be in a single location I'd deploy a remote
Asterisk box and run an IAX trunk between remote and local sites.
That'll save more bandwidth than having a potential 5 individual SIP
sessions running over your link.
Also, with the addition of an analogue card such as the
I would like to setup up a remote office with a half dozen
or so SIP phones connected to an asterisk server via a WAN link. To conserve
bandwidth I would like the phones to be able to re-invite when they call each
other.
The phones will be Polycom, Cisco, or Snom.
I may or may not u
Mikko Suniala wrote:
client A behind Asterisk 1 calls client B behind Asterisk 2, after the
connections have been established, the Asterisks issue reinvites and
they will step out of the media path so that RTP traffic will stream
directly between the clients.
Yes. The media path will collaps
Dear all,
Does reinvite work for a SIP to SIP call if there are more than one
Asterisk between the clients? An example scenario:
A ---> |Asterisk 1| ---> |Asterisk 2| ---> B
client A behind Asterisk 1 calls client B behind Asterisk 2, after the
connections have been established, the Asterisk
Hi,
We're routing SIP calls through Asterisk and we want to
be able to reinvite calls without Asterisk performing
codec conversion.
We've performed the following test:
Asterisk has license for G.729 installed
sip.conf
[general]
context=default
autocreatepeer=yes
disallow=all
allow=alaw
allow
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